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Grandstream Networks, Inc. GXV300x User Manual Page 33 of 54
Firmware 1.2.3.7 Updated: 12/2010
NAT Traversal
(STUN)
This parameter defines whether the NAT traversal mechanism is activated. If activated (by
choosing “Yes”) and a STUN server is also specified, then the GXV300x will behave ac-
cording to the STUN client specification. Using this mode, the embedded STUN client de-
tects what type of NAT/Firewall router used and displayed the result in Status Page.
If the detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the
GXV300x will attempt to use its mapped public IP address and port in all of its SIP and
SDP messages.
If the NAT Traversal field is set to “No, but send keep-alive” with no specified STUN server,
the GXV300x will periodically (every 20 seconds or so) send a blank UDP packet (with no
payload data) to the SIP server to keep the “hole” on the NAT router open.
If the detected NAT is symmetric NAT, then STUN can NOT be used and an Outbound
Proxy or Session Border Controller must be used and this field must be configured to “NO”.
Keep
-
Alive Using SIP O
P-
TIONS
Default is
No
Subscribe for MWI
Default is
No
. When set to “Yes” a SUBSCRIBE for Message Waiting Indication will be
sent periodically. GXV300x supports both synchronized and non-synchronized MWI.
Pro
xy
-
Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Do not configure it unless the server supports this special feature.
SIP Compact Header
Default is No. If set to Yes, the header is sent in compact format. This is helpful in stateless
proxy with multiple codec used as big UDP packet is not router friendly during transit.
Voice Mail User ID
When configured, user can access VM messages by pressing “MSG” button.
This ID is usually the VM portal access number, e.g.: 8500 in Asterisk or 8502 for FWD.
Send DTMF
This parameter specifies the mechanism to transmit DTMF digit.
There are 3 supported modes: in audio which means DTMF is combined in audio signal
(not very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.
Early Dial
Default is
No
. Use only if proxy or server supports 484 response.
Dial Plan Prefix
Sets the prefix added to each dialed number.